--- /dev/null
+<configuration name="switch.conf" description="Core Configuration">
+
+ <cli-keybindings>
+ <key name="1" value="help"/>
+ <key name="2" value="status"/>
+ <key name="3" value="show channels"/>
+ <key name="4" value="show calls"/>
+ <key name="5" value="sofia status"/>
+ <key name="6" value="reloadxml"/>
+ <key name="7" value="console loglevel 0"/>
+ <key name="8" value="console loglevel 7"/>
+ <key name="9" value="sofia status profile internal"/>
+ <key name="10" value="sofia profile internal siptrace on"/>
+ <key name="11" value="sofia profile internal siptrace off"/>
+ <key name="12" value="version"/>
+ </cli-keybindings>
+
+ <default-ptimes>
+ <!-- Set this to override the 20ms assumption of various codecs in the sdp with no ptime defined -->
+ <!-- <codec name="G729" ptime="40"/> -->
+ </default-ptimes>
+
+ <settings>
+ <!-- Colorize the Console -->
+ <param name="colorize-console" value="true"/>
+
+ <!--Include full timestamps in dialplan logs -->
+ <param name="dialplan-timestamps" value="false"/>
+
+ <!-- Run the timer at 20ms by default and drop down as needed unless you set 1m-timer=true which was previous default -->
+ <!-- <param name="1ms-timer" value="true"/> -->
+
+ <!--
+ Set the Switch Name for HA environments.
+ When setting the switch name, it will override the system hostname for all DB and CURL requests
+ allowing cluster environments such as RHCS to have identical FreeSWITCH configurations but run
+ as different hostnames.
+ -->
+ <!-- <param name="switchname" value="freeswitch"/> -->
+ <!-- <param name="cpu-idle-smoothing-depth" value="30"/> -->
+
+
+ <!-- Maximum number of simultaneous DB handles open -->
+ <param name="max-db-handles" value="50"/>
+ <!-- Maximum number of seconds to wait for a new DB handle before failing -->
+ <param name="db-handle-timeout" value="10"/>
+
+ <!-- Minimum idle CPU before refusing calls -->
+ <!-- <param name="min-idle-cpu" value="25"/> -->
+
+ <!-- Interval between heartbeat events -->
+ <!-- <param name="event-heartbeat-interval" value="20"/> -->
+
+ <!--
+ Max number of sessions to allow at any given time.
+
+ NOTICE: If you're driving 28 T1's in a single box you should set this to 644*2 or 1288
+ this will ensure you're able to use the entire DS3 without a problem. Otherwise you'll
+ be 144 channels short of always filling that DS3 up which can translate into waste.
+ -->
+ <param name="max-sessions" value="1000"/>
+ <!--Most channels to create per second -->
+ <param name="sessions-per-second" value="30"/>
+ <!-- Default Global Log Level - value is one of debug,info,notice,warning,err,crit,alert -->
+ <param name="loglevel" value="debug"/>
+
+ <!-- Set the core DEBUG level (0-10) -->
+ <!-- <param name="debug-level" value="10"/> -->
+
+ <!-- SQL Buffer length within rage of 32k to 10m -->
+ <!-- <param name="sql-buffer-len" value="1m"/> -->
+ <!-- Maximum SQL Buffer length must be greater than sql-buffer-len -->
+ <!-- <param name="max-sql-buffer-len" value="2m"/> -->
+
+ <!--
+ The min-dtmf-duration specifies the minimum DTMF duration to use on
+ outgoing events. Events shorter than this will be increased in duration
+ to match min_dtmf_duration. You cannot configure a dtmf duration on a
+ profile that is less than this setting. You may increase this value,
+ but cannot set it lower than 400. This value cannot exceed
+ max-dtmf-duration. -->
+ <!-- <param name="min-dtmf-duration" value="400"/> -->
+
+ <!--
+ The max-dtmf-duration caps the playout of a DTMF event at the specified
+ duration. Events exceeding this duration will be truncated to this
+ duration. You cannot configure a duration on a profile that exceeds
+ this setting. This setting can be lowered, but cannot exceed 192000.
+ This setting cannot be set lower than min_dtmf_duration. -->
+ <!-- <param name="max-dtmf-duration" value="192000"/> -->
+
+ <!--
+ The default_dtmf_duration specifies the DTMF duration to use on
+ originated DTMF events or on events that are received without a
+ duration specified. This value can be increased or lowered. This
+ value is lower-bounded by min_dtmf_duration and upper-bounded by
+ max-dtmf-duration\. -->
+ <!-- <param name="default-dtmf-duration" value="2000"/> -->
+
+ <!--
+ If you want to send out voicemail notifications via Windows you'll need to change the mailer-app
+ variable to the setting below:
+
+ <param name="mailer-app" value="msmtp"/>
+
+ Do not change mailer-app-args.
+ You will also need to download a sendmail clone for Windows (msmtp). This version works without issue:
+ http://msmtp.sourceforge.net/index.html. Download and copy the .exe to %winddir%\system32.
+ You'll need to create a small config file for smtp credentials (host name, authentication, tls, etc.) in
+ %USERPROFILE%\Application Data\ called "msmtprc.txt". Below is a sample copy of this file:
+
+ ###################################
+ # The SMTP server of the provider.
+ account provider
+ host smtp.myisp.com
+ from john@myisp.com
+ auth login
+ user johndoe
+ password mypassword
+
+ # Set a default account
+ account default : provider
+ ###################################
+
+ -->
+
+ <param name="mailer-app" value="sendmail"/>
+ <param name="mailer-app-args" value="-t"/>
+ <param name="dump-cores" value="yes"/>
+
+ <!-- Enable verbose channel events to include every detail about a channel on every event -->
+ <!-- <param name="verbose-channel-events" value="no"/> -->
+
+ <!-- Enable clock nanosleep -->
+ <!-- <param name="enable-clock-nanosleep" value="true"/> -->
+
+ <!-- Enable monotonic timing -->
+ <!-- <param name="enable-monotonic-timing" value="true"/> -->
+
+ <!-- NEEDS DOCUMENTATION -->
+ <!-- <param name="enable-softtimer-timerfd" value="true"/> -->
+ <!-- <param name="enable-cond-yield" value="true"/> -->
+ <!-- <param name="enable-timer-matrix" value="true"/> -->
+ <!-- <param name="threaded-system-exec" value="true"/> -->
+ <!-- <param name="tipping-point" value="0"/> -->
+ <!-- <param name="timer-affinity" value="disabled"/> -->
+ <!-- NEEDS DOCUMENTATION -->
+
+ <!-- RTP port range -->
+ <param name="rtp-start-port" value="<%= @freeswitch_rtp_start_port %>"/>
+ <param name="rtp-end-port" value="<%= @freeswitch_rtp_end_port %>"/>
+
+ <!-- Test each port to make sure it is not in use by some other process before allocating it to RTP -->
+ <!-- <param name="rtp-port-usage-robustness" value="true"/> -->
+
+ <param name="rtp-enable-zrtp" value="false"/>
+
+ <!--
+ Store encryption keys for secure media in channel variables and call CDRs. Default: false.
+ WARNING: If true, anyone with CDR access can decrypt secure media!
+ -->
+ <!-- <param name="rtp-retain-crypto-keys" value="true"/> -->
+
+ <!--
+ Native PostgreSQL support was removed from the FreeSWITCH Core!
+ =================================
+ NOTICE: You MUST enable mod_pgsql
+ =================================
+ According to https://www.postgresql.org/docs/9.6/libpq-connect.html#LIBPQ-CONNSTRING
+ There are two accepted formats for connection strings supported by the libpq library:
+ * For plain keyword = value strings use pgsql://
+ pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE'
+ * For RFC 3986 URIs use postgresql:// or postgres://
+ postgresql://
+ postgresql://localhost
+ postgresql://localhost:5433
+ postgresql://localhost/mydb
+ postgresql://user@localhost
+ postgresql://user:secret@localhost
+ postgresql://other@localhost/otherdb?connect_timeout=10&application_name=myapp
+ postgresql:///mydb?host=localhost&port=5433
+ -->
+ <!-- <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE'" /> -->
+ <!-- <param name="core-db-dsn" value="postgresql://freeswitch:@127.0.0.1/freeswitch?options=-c%20client_min_messages%3DNOTICE" /> -->
+ <!-- <param name="core-db-dsn" value="mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=pass;" /> -->
+ <!-- <param name="core-db-dsn" value="dsn:username:password" /> -->
+ <!--
+ Allow to specify the sqlite db at a different location (In this example, move it to ramdrive for
+ better performance on most linux distro (note, you loose the data if you reboot))
+ -->
+ <!-- <param name="core-db-name" value="/dev/shm/core.db" /> -->
+
+ <!-- The system will create all the db schemas automatically, set this to false to avoid this behaviour -->
+ <!-- <param name="auto-create-schemas" value="true"/> -->
+ <!-- <param name="auto-clear-sql" value="true"/> -->
+ <!-- <param name="enable-early-hangup" value="true"/> -->
+
+ <!-- <param name="core-dbtype" value="MSSQL"/> -->
+
+ <!-- Allow multiple registrations to the same account in the central registration table -->
+ <!-- <param name="multiple-registrations" value="true"/> -->
+
+ <!-- <param name="max-audio-channels" value="2"/> -->
+
+ </settings>
+
+</configuration>
+
--- /dev/null
+<include>
+ <!-- Preprocessor Variables
+ These are introduced when configuration strings must be consistent across modules.
+ NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead.
+
+ WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
+
+ YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any
+ toll fraud in the future. It's your responsibility to secure your own system.
+
+ This default config is used to demonstrate the feature set of FreeSWITCH.
+
+ WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
+ -->
+ <X-PRE-PROCESS cmd="set" data="default_password=<%= @freeswitch_password %>"/>
+ <!-- Did you change it yet? -->
+ <!--
+ The following variables are set dynamically - calculated if possible by freeswitch - and
+ are available to the config as $${variable}. You can see their calculated value via fs_cli
+ by entering eval $${variable}
+
+ hostname
+ local_ip_v4
+ local_mask_v4
+ local_ip_v6
+ switch_serial
+ base_dir
+ recordings_dir
+ sound_prefix
+ sounds_dir
+ conf_dir
+ log_dir
+ run_dir
+ db_dir
+ mod_dir
+ htdocs_dir
+ script_dir
+ temp_dir
+ grammar_dir
+ certs_dir
+ storage_dir
+ cache_dir
+ core_uuid
+ zrtp_enabled
+ nat_public_addr
+ nat_private_addr
+ nat_type
+
+ -->
+
+
+ <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
+ <!--<Z-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/allison"/> -->
+
+ <!--
+ This setting is what sets the default domain FreeSWITCH will use if all else fails.
+
+ FreeSWICH will default to $${local_ip_v4} unless changed. Changing this setting does
+ affect the sip authentication. Please review conf/directory/default.xml for more
+ information on this topic.
+ -->
+ <X-PRE-PROCESS cmd="set" data="domain=sip.pdp7.net"/>
+ <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
+ <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
+ <X-PRE-PROCESS cmd="set" data="use_profile=external"/>
+ <X-PRE-PROCESS cmd="set" data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH"/>
+ <!--
+ Enable ZRTP globally you can override this on a per channel basis
+
+ http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)
+ -->
+ <X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
+ <!--
+ NOTICE: When using SRTP it's critical that you do not offer or accept
+ variable bit rate codecs, doing so would leak information and possibly
+ compromise your SRTP stream. (FS-6404)
+
+ Supported SRTP Crypto Suites:
+
+ AEAD_AES_256_GCM_8
+ ____________________________________________________________________________
+ This algorithm is identical to AEAD_AES_256_GCM (see Section 5.2 of
+ [RFC5116]), except that the tag length, t, is 8, and an
+ authentication tag with a length of 8 octets (64 bits) is used.
+ An AEAD_AES_256_GCM_8 ciphertext is exactly 8 octets longer than its
+ corresponding plaintext.
+
+
+ AEAD_AES_128_GCM_8
+ ____________________________________________________________________________
+ This algorithm is identical to AEAD_AES_128_GCM (see Section 5.1 of
+ [RFC5116]), except that the tag length, t, is 8, and an
+ authentication tag with a length of 8 octets (64 bits) is used.
+ An AEAD_AES_128_GCM_8 ciphertext is exactly 8 octets longer than its
+ corresponding plaintext.
+
+
+ AES_CM_256_HMAC_SHA1_80 | AES_CM_192_HMAC_SHA1_80 | AES_CM_128_HMAC_SHA1_80
+ ____________________________________________________________________________
+ AES_CM_128_HMAC_SHA1_80 is the SRTP default AES Counter Mode cipher
+ and HMAC-SHA1 message authentication with an 80-bit authentication
+ tag. The master-key length is 128 bits and has a default lifetime of
+ a maximum of 2^48 SRTP packets or 2^31 SRTCP packets, whichever comes
+ first.
+
+
+ AES_CM_256_HMAC_SHA1_32 | AES_CM_192_HMAC_SHA1_32 | AES_CM_128_HMAC_SHA1_32
+ ____________________________________________________________________________
+ This crypto-suite is identical to AES_CM_128_HMAC_SHA1_80 except that
+ the authentication tag is 32 bits. The length of the base64-decoded key and
+ salt value for this crypto-suite MUST be 30 octets i.e., 240 bits; otherwise,
+ the crypto attribute is considered invalid.
+
+
+ AES_CM_128_NULL_AUTH
+ ____________________________________________________________________________
+ The SRTP default cipher (AES-128 Counter Mode), but to use no authentication
+ method. This policy is NOT RECOMMENDED unless it is unavoidable; see
+ Section 7.5 of [RFC3711].
+
+
+ SRTP variables that modify behaviors based on direction/leg:
+
+ rtp_secure_media
+ ____________________________________________________________________________
+ possible values:
+ mandatory - Accept/Offer SAVP negotiation ONLY
+ optional - Accept/Offer SAVP/AVP with SAVP preferred
+ forbidden - More useful for inbound to deny SAVP negotiation
+ false - implies forbidden
+ true - implies mandatory
+
+ default if not set is accept SAVP inbound if offered.
+
+
+ rtp_secure_media_inbound | rtp_secure_media_outbound
+ ____________________________________________________________________________
+ This is the same as rtp_secure_media, but would apply to either inbound
+ or outbound offers specifically.
+
+
+ How to specify crypto suites:
+ ____________________________________________________________________________
+ By default without specifying any crypto suites FreeSWITCH will offer
+ crypto suites from strongest to weakest accepting the strongest each
+ endpoint has in common. If you wish to force specific crypto suites you
+ can do so by appending the suites in a comma separated list in the order
+ that you wish to offer them in.
+
+ Examples:
+
+ rtp_secure_media=mandatory:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
+ rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
+ rtp_secure_media=optional:AES_CM_256_HMAC_SHA1_80
+ rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80
+
+ Additionally you can narrow this down on either inbound or outbound by
+ specifying as so:
+
+ rtp_secure_media_inbound=true:AEAD_AES_256_GCM_8
+ rtp_secure_media_inbound=mandatory:AEAD_AES_256_GCM_8
+ rtp_secure_media_outbound=true:AEAD_AES_128_GCM_8
+ rtp_secure_media_outbound=optional:AEAD_AES_128_GCM_8
+
+
+ rtp_secure_media_suites
+ ____________________________________________________________________________
+ Optionaly you can use rtp_secure_media_suites to dictate the suite list
+ and only use rtp_secure_media=[optional|mandatory|false|true] without having
+ to dictate the suite list with the rtp_secure_media* variables.
+ -->
+ <!--
+ Examples of codec options: (module must be compiled and loaded)
+
+ codecname[@8000h|16000h|32000h[@XXi]]
+
+ XX is the frame size must be multples allowed for the codec
+ FreeSWITCH can support 10-120ms on some codecs.
+ We do not support exceeding the MTU of the RTP packet.
+
+
+ iLBC@30i - iLBC using mode=30 which will win in all cases.
+ DVI4@8000h@20i - IMA ADPCM 8kHz using 20ms ptime. (multiples of 10)
+ DVI4@16000h@40i - IMA ADPCM 16kHz using 40ms ptime. (multiples of 10)
+ speex@8000h@20i - Speex 8kHz using 20ms ptime.
+ speex@16000h@20i - Speex 16kHz using 20ms ptime.
+ speex@32000h@20i - Speex 32kHz using 20ms ptime.
+ BV16 - BroadVoice 16kb/s narrowband, 8kHz
+ BV32 - BroadVoice 32kb/s wideband, 16kHz
+ G7221@16000h - G722.1 16kHz (aka Siren 7)
+ G7221@32000h - G722.1C 32kHz (aka Siren 14)
+ CELT@32000h - CELT 32kHz, only 10ms supported
+ CELT@48000h - CELT 48kHz, only 10ms supported
+ GSM@40i - GSM 8kHz using 40ms ptime. (GSM is done in multiples of 20, Default is 20ms)
+ G722 - G722 16kHz using default 20ms ptime. (multiples of 10)
+ PCMU - G711 8kHz ulaw using default 20ms ptime. (multiples of 10)
+ PCMA - G711 8kHz alaw using default 20ms ptime. (multiples of 10)
+ G726-16 - G726 16kbit adpcm using default 20ms ptime. (multiples of 10)
+ G726-24 - G726 24kbit adpcm using default 20ms ptime. (multiples of 10)
+ G726-32 - G726 32kbit adpcm using default 20ms ptime. (multiples of 10)
+ G726-40 - G726 40kbit adpcm using default 20ms ptime. (multiples of 10)
+ AAL2-G726-16 - Same as G726-16 but using AAL2 packing. (multiples of 10)
+ AAL2-G726-24 - Same as G726-24 but using AAL2 packing. (multiples of 10)
+ AAL2-G726-32 - Same as G726-32 but using AAL2 packing. (multiples of 10)
+ AAL2-G726-40 - Same as G726-40 but using AAL2 packing. (multiples of 10)
+ LPC - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH)
+ L16 - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly.
+
+ These are the passthru audio codecs:
+
+ G729 - G729 in passthru mode. (mod_g729)
+ G723 - G723.1 in passthru mode. (mod_g723_1)
+ AMR - AMR in passthru mode. (mod_amr)
+
+ These are the passthru video codecs: (mod_h26x)
+
+ H261 - H.261 Video
+ H263 - H.263 Video
+ H263-1998 - H.263-1998 Video
+ H263-2000 - H.263-2000 Video
+ H264 - H.264 Video
+
+ RTP Dynamic Payload Numbers currently used in FreeSWITCH and their purpose.
+
+ 96 - AMR
+ 97 - iLBC (30)
+ 98 - iLBC (20)
+ 99 - Speex 8kHz, 16kHz, 32kHz
+ 100 -
+ 101 - telephone-event
+ 102 -
+ 103 -
+ 104 -
+ 105 -
+ 106 - BV16
+ 107 - G722.1 (16kHz)
+ 108 -
+ 109 -
+ 110 -
+ 111 -
+ 112 -
+ 113 -
+ 114 - CELT 32kHz, 48kHz
+ 115 - G722.1C (32kHz)
+ 116 -
+ 117 - SILK 8kHz
+ 118 - SILK 12kHz
+ 119 - SILK 16kHz
+ 120 - SILK 24kHz
+ 121 - AAL2-G726-40 && G726-40
+ 122 - AAL2-G726-32 && G726-32
+ 123 - AAL2-G726-24 && G726-24
+ 124 - AAL2-G726-16 && G726-16
+ 125 -
+ 126 -
+ 127 - BV32
+
+ -->
+ <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,H264,VP8"/>
+ <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,H264,VP8"/>
+
+ <!--
+ xmpp_client_profile and xmpp_server_profile
+ xmpp_client_profile can be any string.
+ xmpp_server_profile is appended to "dingaling_" to form the database name
+ containing the "subscriptions" table.
+ used by: dingaling.conf.xml enum.conf.xml
+ -->
+
+ <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
+ <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
+ <!--
+ THIS IS ONLY USED FOR DINGALING
+
+ bind_server_ip
+
+ Can be an ip address, a dns name, or "auto".
+ This determines an ip address available on this host to bind.
+ If you are separating RTP and SIP traffic, you will want to have
+ use different addresses where this variable appears.
+ Used by: dingaling.conf.xml
+ -->
+ <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
+
+ <!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
+
+ If you're going to load test FreeSWITCH please input real IP addresses
+ for external_rtp_ip and external_sip_ip
+ -->
+
+ <!-- external_rtp_ip
+ Can be an one of:
+ ip address: "12.34.56.78"
+ a stun server lookup: "stun:stun.server.com"
+ a DNS name: "host:host.server.com"
+ where fs.mydomain.com is a DNS A record-useful when fs is on
+ a dynamic IP address, and uses a dynamic DNS updater.
+ If unspecified, the bind_server_ip value is used.
+ Used by: sofia.conf.xml dingaling.conf.xml
+ -->
+ <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=<%= @freeswitch_address %>"/>
+
+ <!-- external_sip_ip
+ Used as the public IP address for SDP.
+ Can be an one of:
+ ip address: "12.34.56.78"
+ a stun server lookup: "stun:stun.server.com"
+ a DNS name: "host:host.server.com"
+ where fs.mydomain.com is a DNS A record-useful when fs is on
+ a dynamic IP address, and uses a dynamic DNS updater.
+ If unspecified, the bind_server_ip value is used.
+ Used by: sofia.conf.xml dingaling.conf.xml
+ -->
+ <X-PRE-PROCESS cmd="stun-set" data="external_sip_ip=<%= @freeswitch_address %>"/>
+
+ <!-- unroll-loops
+ Used to turn on sip loopback unrolling.
+ -->
+ <X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>
+
+ <!-- outbound_caller_id and outbound_caller_name
+ The caller ID telephone number we should use when calling out.
+ Used by: conference.conf.xml and user directory for default
+ outbound callerid name and number.
+ -->
+ <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
+ <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
+
+ <!-- various debug and defaults -->
+ <X-PRE-PROCESS cmd="set" data="call_debug=false"/>
+ <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
+ <X-PRE-PROCESS cmd="set" data="default_areacode=918"/>
+ <X-PRE-PROCESS cmd="set" data="default_country=US"/>
+
+ <!-- if false or undefined, the destination number is included in presence NOTIFY dm:note.
+ if true, the destination number is not included -->
+ <X-PRE-PROCESS cmd="set" data="presence_privacy=false"/>
+
+ <X-PRE-PROCESS cmd="set" data="au-ring=%(400,200,383,417);%(400,2000,383,417)"/>
+ <X-PRE-PROCESS cmd="set" data="be-ring=%(1000,3000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="ca-ring=%(2000,4000,440,480)"/>
+ <X-PRE-PROCESS cmd="set" data="cn-ring=%(1000,4000,450)"/>
+ <X-PRE-PROCESS cmd="set" data="cy-ring=%(1500,3000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="cz-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="de-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="dk-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="dz-ring=%(1500,3500,425)"/>
+ <X-PRE-PROCESS cmd="set" data="eg-ring=%(2000,1000,475,375)"/>
+ <X-PRE-PROCESS cmd="set" data="es-ring=%(1500,3000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="fi-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440)"/>
+ <X-PRE-PROCESS cmd="set" data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/>
+ <X-PRE-PROCESS cmd="set" data="hu-ring=%(1250,3750,425)"/>
+ <X-PRE-PROCESS cmd="set" data="il-ring=%(1000,3000,400)"/>
+ <X-PRE-PROCESS cmd="set" data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/>
+ <X-PRE-PROCESS cmd="set" data="jp-ring=%(1000,2000,420,380)"/>
+ <X-PRE-PROCESS cmd="set" data="ko-ring=%(1000,2000,440,480)"/>
+ <X-PRE-PROCESS cmd="set" data="pk-ring=%(1000,2000,400)"/>
+ <X-PRE-PROCESS cmd="set" data="pl-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="ro-ring=%(1850,4150,475,425)"/>
+ <X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425)"/>
+ <X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425)"/>
+ <X-PRE-PROCESS cmd="set" data="sa-ring=%(1200,4600,425)"/>
+ <X-PRE-PROCESS cmd="set" data="tr-ring=%(2000,4000,450)"/>
+ <X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/>
+ <X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440,480)"/>
+ <X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
+ <X-PRE-PROCESS cmd="set" data="beep=%(1000,0,640)"/>
+ <X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
+
+ <!--
+ Digits Dialed filter: (FS-6940)
+
+ The digits stream may contain valid credit card numbers or social security numbers, These digit
+ filters will allow you to make a valant effort to stamp out sensitive information for
+ PCI/HIPPA compliance. (see xml_cdr dialed_digits)
+
+ df_us_ssn = US Social Security Number pattern
+ df_us_luhn = Visa, MasterCard, American Express, Diners Club, Discover and JCB
+ -->
+ <X-PRE-PROCESS cmd="set" data="df_us_ssn=(?!219099999|078051120)(?!666|000|9\d{2})\d{3}(?!00)\d{2}(?!0{4})\d{4}"/>
+ <X-PRE-PROCESS cmd="set" data="df_luhn=?:4[0-9]{12}(?:[0-9]{3})?|5[1-5][0-9]{14}|3[47][0-9]{13}|3(?:0[0-5]|[68][0-9])[0-9]{11}|6(?:011|5[0-9]{2})[0-9]{12}|(?:2131|1800|35\d{3})\d{11}"/>
+ <!-- change XX to X below to enable -->
+ <XX-PRE-PROCESS cmd="set" data="digits_dialed_filter=(($${df_luhn})|($${df_us_ssn}))"/>
+
+ <!--
+ Setting up your default sip provider is easy.
+ Below are some values that should work in most cases.
+
+ These are for conf/directory/default/example.com.xml
+ -->
+ <X-PRE-PROCESS cmd="set" data="default_provider=example.com"/>
+ <X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/>
+ <X-PRE-PROCESS cmd="set" data="default_provider_password=password"/>
+ <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/>
+ <!-- true or false -->
+ <X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>
+ <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
+
+ <!--
+ SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls
+
+ valid options: sslv2,sslv3,sslv23,tlsv1,tlsv1.1,tlsv1.2
+
+ default: tlsv1,tlsv1.1,tlsv1.2
+ -->
+ <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1,tlsv1.1,tlsv1.2"/>
+
+ <!--
+ TLS cipher suite: default ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH
+
+ The actual ciphers supported will change per platform.
+
+ openssl ciphers -v 'ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH'
+
+ Will show you what is available in your verion of openssl.
+ -->
+ <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/>
+
+ <!-- Internal SIP Profile -->
+ <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
+ <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
+ <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
+ <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
+
+ <!-- External SIP Profile -->
+ <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
+ <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
+ <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
+ <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
+
+ <!-- Video Settings -->
+ <!-- Setting the max bandwdith -->
+ <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=3mb"/>
+ <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=3mb"/>
+
+ <!-- WebRTC Video -->
+ <!-- Suppress CNG for WebRTC Audio -->
+ <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
+ <!-- Enable liberal DTMF for those that can't get it right -->
+ <X-PRE-PROCESS cmd="set" data="rtp_liberal_dtmf=true"/>
+ <!-- Helps with WebRTC Audio -->
+
+ <!-- Stock Video Avatars -->
+ <X-PRE-PROCESS cmd="set" data="video_mute_png=$${images_dir}/default-mute.png"/>
+ <X-PRE-PROCESS cmd="set" data="video_no_avatar_png=$${images_dir}/default-avatar.png"/>
+
+</include>
projects / alex.git / commitdiff